Abstract: Methods and systems include sending recording data of a call to a first server and a second server, wherein the recording data includes a first voice of a first participant of the call and a second voice of a second participant of the call; receiving, from the first server, a first emotion score representing a degree of a first emotion associated with the first voice, and a second emotion score representing a degree of a second emotion associated with the first voice; receiving, from the second server, a first sentiment score, a second sentiment score, and a third sentiment score; determining a quality score and classification data for the recording data based on the first emotion score, the second emotion score, the first sentiment score, the second sentiment score, and the third sentiment score; and outputting the quality score and the classification data for visualization of the recording data.

Abstract: A computerized communication device has a display screen, a mechanism for a user to select words or phrases displayed on the display screen, and software executing from a non-transitory physical medium, the software providing a function for providing audio signal output in a connected voice-telephone call from the text words or phrases selected by a user.

Abstract: A communications system may include at least one mobile wireless communications device, and a wireless communications network for sending text messages thereto. More particularly, the at least one mobile wireless communications device may include a wireless transceiver and a controller for cooperating therewith for receiving text messages from the wireless communications network. It may further include a headset output connected to the controller. The controller may be for switching between a normal message mode and an audio message mode based upon a connection between the headset output and a headset. Moreover, when in the audio message mode, the controller may output at least one audio message including speech generated from at least one of the received text messages via the headset output.

Abstract: A communications system may include at least one mobile wireless communications device, and a wireless communications network for sending text messages thereto. More particularly, the at least one mobile wireless communications device may include a wireless transceiver and a controller for cooperating therewith for receiving text messages from the wireless communications network. It may further include a headset output connected to the controller. The controller may be for switching between a normal message mode and an audio message mode based upon a connection between the headset output and a headset. Moreover, when in the audio message mode, the controller may output at least one audio message including speech generated from at least one of the received text messages via the headset output.

Abstract: A unified communication system is disclosed that allows a variety of end point types to participate in a communication event using a common, unified communication system. In some implementations, a calling party interacts with a client application residing on an endpoint to make a communication request to another endpoint. A communication event manager residing in the unified communication system selects a script from a repository of scripts based on the communication event and the capabilities of the endpoints. A communication event execution engine receives a user profile associated with at least one of the endpoints. The user profile can be configured by the user to describe the user's preferences for how the communication should be processed by the unified communication system.

Abstract: A spell checking system includes a letter spelling engine. The letter spelling engine is configured to select a plurality of candidate letter target strings that closely match a misspelled source string. The spell checking system includes a phoneme spelling engine. The phoneme spelling engine is configured to select a plurality of candidate phoneme target strings that closely match the misspelled source string. A ranker module is configured to combine the candidate letter target strings and the candidate phoneme target strings into a combined list of candidate target strings. The ranker module is also configured to rank the list of candidate target strings to provide a list of best candidate target strings for the misspelled source string.

Abstract: Data received by a receiver is processed at a sampling rate indicated manually or automatically in the receiver. The sampling rate of the received data is controlled in accordance with the processing rate. The sampling rate controlled data is then processed so as to convert its frequency distribution to that the received data originally had.

Abstract: The present research can decrease the amount of computation and enhance speech quality by using a global pulse replacement method in a fixed codebook search. The fixed codebook search method in a speech encoder based upon global pulse replacement, includes the steps of: (a) computing absolute values of the pulse-position likelihood-estimator vectors; (b) temporarily obtaining a codebook vector; (c) computing a mathematical equation by replacing a pulse; (d) determining whether a value computed based upon the mathematical equation is increased after pulse replacement; (e) obtaining a new codebook vector by replacing the pulse; and (f) maintaining a previous codebook vector.

Abstract: A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal; and from the speech signal, deriving a spectral envelope signal representing the modelled filter and a remaining signal representing the modelled source. At intervals during the encoding, the method further comprises determining a period between portions of the remaining signal having a degree of repetition and determining a correlation between said portions based on that period, thus producing a respective vector of the correlation for each interval. Once every number of said intervals, the method further comprises selecting a codebook from a plurality of codebooks for quantizing the vectors, quantizing the vectors of that number of intervals according to the selected codebook, and transmitting the quantized vectors along with an indication of the selected codebook.

Abstract: The present invention relates to creating a phonetic index of phonemes from an audio segment that includes speech content from multiple sources. The phonemes in the phonetic index are directly or indirectly associated with the corresponding source of the speech from which the phonemes were derived. By associating the phonemes with a corresponding source, the phonetic index of speech content from multiple sources may be searched based on phonetic content as well as the corresponding source.

Abstract: The invention relates to compression coding and/or decoding of digital signals, in particular by vector variable-rate quantisation defining a variable resolution. For this purpose an impulsion dictionary comprises: for a given dimension, increasing resolution dictionaries imbricated into each other and, for a given dimension, a union of: a totality (D?i<N>) of code-vectors produced, by inserting elements taken in a final set (A) into smaller dimension code-vectors according to a final set of predetermined insertion rules (F1) and a second totality of code-vectors (Y?) which are not obtainable by insertion into the smaller dimension code vectors according to said set of the insertion rules.

Abstract: Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3.

Abstract: The present invention relates to speech coding in wireless and wireline communication systems. The present invention provides a method of saving bandwidth by a controlled dropping of speech frames at an encoder in a sending communication device. The dropping is controlled in a manner to minimize the effects on the speech quality after the decoding in the receiving communication device, by assuring that the state mismatch between the encoder and the decoder is removed or at least significantly reduced. This is achieved by letting the encoder run an ECU algorithm with a similar behavior as the one running in the decoder in the receiving communication device.

Abstract: Call management methods and systems for use in a device having telecommunication capability. At least one communication event comprising a phone number, first communication content and communication time is provided. At the communication time, a call is made according to the phone number. A first communication voice is generated using a text-to-speech technology according to the first communication content, and provided to at least one participant of the call.

Abstract: A mobile communication terminal having a phonebook replays a stored message when an entry of the phonebook is selected using a speed dial number and the stored message is associated with the selected phonebook entry. The message may be the spoken name of the person associated with the selected phonebook entry thereby enabling the terminal user to have an audio confirmation of the party being called. The message may be recorded by the terminal microphone or may be captured from a connected call. Alternatively the message may be a text message that is synthesized to a voice message by a text-to-speech function.

Abstract: Data processing system and computer implemented method for obtaining linguistic image labels and populating linguistic image label entries are disclosed. According to one embodiment, a method comprises creating a first data from an image that includes descriptive information of the image. A linguistic image label is populated that includes a first field and a second field wherein the first field contains first data representing a pixel region of a digital image and the second field contains second data representing a visual appearance of the pixel region.